In order to construct a call system, such as a vehicle-mounted handsfree system, in a high-noise environment or an environment where a plurality of signal sources exist, a technique of separating and extracting only a signal of a specific signal source (speaker) is required. A beamformer is provided as one example of this technique. The beamformer enhances a signal in a target direction by adding signals of multiple channels provided by a microarray, and includes a fixed beamformer and an adaptive beamformer.
The simplest fixed beamformer is based on Delay and Sum, and is comprised of microphones 901 and 902 of two channels, a signal delaying unit 903, and a delay summing unit 904, as shown in FIG. 6. While this Delay and Sum generally requires only a small amount of computations, a problem with the Delay and Sum is that when it is difficult to use a large number of microphones, such as when the Delay and Sum is used for vehicle-mounted use, a sidelobe is large, the method is not effective in a reverberation environment, and adequate directivity is not acquired for a low frequency region.
In order to improve the directivity in a low frequency region, it is necessary to lengthen the array length of the entire microphone array. For example, when a main lobe is made to have directivity of about ±10 degrees for a 1000-Hz sound, it is necessary to make the array length be about 2 m. A further problem is that when the array length is increased by simply lengthening the intervals of the microphone array, a grating lobe occurs in a direction other than the target direction, and the directivity degrades (refer to nonpatent reference 1). Therefore, another problem is that in order to suppress the grating lobe and maintain the directivity in the low frequency region, it is necessary to arrange a large number of microphones densely, and hence the fixed beamformer costs highly.
In contrast with this, the adaptive beamformer is based on a method of forming directivity in such a way that a noise sound source is located in a blind spot while holding the sensitivity in a target direction at a constant level, and is effective also for a low frequency region and can carryout noise suppression in a reverberation environment. Although there are various methods for use in the adaptive beamformer, there is a generalized sidelobe canceller (GSC) as one of methods which can be assumed to be an extension of the Delay and Sum. The generalized sidelobe canceller is a beamformer that suppresses noise by using a fixed beamformer and an adaptive filter, and a typical Griffith-Jim type GSC using microphones of two channels is constructed as shown in FIG. 7. This GSC is comprised of microphones 901 and 902 of two channels, a signal delaying unit 903, a delay summing unit 904, a target sound blocker 905, and an adaptive filter 906, and the target sound blocker 905 carries out subtraction-type beamforming based on a subtraction of microphone signals. The adaptive filter 906 estimates a noise component by using an output of the target sound blocker 905, and determines a difference with an output of the delay summing unit 904.
It is considered that only a noise component in which a target signal is subtracted remains in the output of the subtraction-type beamformer, and the noise component can be removed from the result of the Delay and Sum by applying the output as an input to the adaptive filter. A problem is, however, that only the simple subtraction cannot sufficiently remove the target signal in many cases, and the adaptive filter cannot sufficiently remove the noise, but ends up removing the target signal.
As a measure against this problem, in a device disclosed by patent reference 1, a target sound blocker is constructed of an adaptive filter using an output of a fixed beamformer and microphone inputs, and is constructed in such a way as to remove a target signal from each of the microphone inputs. Because a signal from which the target sound is removed more sufficiently as compared with a simple subtraction-type beamformer is acquired, the noise suppression performance of the adaptive filter in the next stage can be improved.